Original author(s) | SIPphone |
---|---|
Developer(s) | |
Initial release | July 2005 (as SIPphone) |
Final release | 4.0.5.400 (Windows) November 12, 2009 , 4.0.0.269 (Mac) September 23, 2009 , 3.1.0.79 (Linux) November 29, 2007 |
Operating system | Mac OS X, Linux, Windows, Internet Tablet OS, Symbian |
Type | Peer-to-peer internet telephony |
License | Freeware |
Website | www |
Gizmo5 (formerly known as Gizmo Project and SIPphone) was a voice over IP communications network and a proprietary freeware soft phone for that network. On November 12, 2009, Google announced that it had acquired Gizmo5. [1] On March 4, 2011, Google announced that the service would be discontinued as of April 3, 2011. [2]
The Gizmo5 network used open standards for call management, Session Initiation Protocol (SIP) and Extensible Messaging and Presence Protocol (XMPP). [3] However, the Gizmo5 client application was proprietary software and used several proprietary codecs, including GIPS and Internet Speech Audio Codec (iSAC).[ citation needed ]
Gizmo Project was founded by Michael Robertson and his company SIPphone. [4]
On November 12, 2009, Google announced that it had acquired Gizmo5 [1] for a reported $30 million in cash. Prior to this acquisition, Gizmo5 had a working relationship with GrandCentral (now Google Voice) for years.[ citation needed ] Upon announcement, Gizmo5 suspended new signups until a Google relaunch. [5] Google was also dogfooding a Google Voice desktop client based on Gizmo5, branded as Gizmo5 by Google. [6]
On April 3, 2011, Google shut down Gizmo5 and recommended users to use Google Talk instead. [7]
Gizmo5 was based on the Session Initiation Protocol and could interoperate with other SIP-based networks directly, including the public switched telephone network. The latter required the Gizmo5 service features CallOut and CallIn. CallOut was available at a fee, whereas CallIn and calls to other VoIP users were generally free of cost. Gizmo5 also used encryption (Secure Real-time Transport Protocol) for network calls and worked well with Phil Zimmermann's Zfone [8] security features.[ citation needed ]
Gizmo5 supported the following Codecs:
Version 4.0 of the Gizmo5 softphone offered video calls. Gizmo5 also offered smartphone version.
As of July 20, 2009, Gizmo5 was the only SIP service that could be used with Google Voice directly (without requiring a U.S. based phone number).
The text chat function of Gizmo5 utilized the Extensible Messaging and Presence Protocol (XMPP) protocol. [9] Users were addressed by an identification string in the format of username@chat.gizmoproject.com.
An earlier incarnation of the service was PhoneGaim, a free software VoIP system based on the Pidgin instant messaging software and the SIP protocol handling of the Linphone VoIP software, but restricted to using (only) the SIPphone service. It is available under the GNU General Public License and sponsored by Linspire.
Gizmo5 supported outbound caller line identification in the United States. [10]
Gizmo5 provided a free voicemail service. [11]
Gizmo5 allowed paying subscribers of LiveJournal to place voiceposts if they are unable to use the voicepost telephone lines provided by the website. [12]
The Gizmo5 mobile phone application used the phone's carrier voice network for all calls. The service called the phone numbers of both parties and bridged the call. On mobile phones that support SIP applications, calls may be placed over WiFi or 3G. In the case of WiFi, calls to Gizmo5 users were free, and calls to the public switched telephone network were charged Gizmo5 Call Out credit. On 3G, additional costs would apply depending on the user's data plan.[ citation needed ]
On August 26, 2010, Gmail accounts with Google voice were given a function to make and receive calls. Google Voice product manager, Vincent Paquet, confirmed that this function was added through the help of the technology received after the Gizmo5 acquisition. [13]
On Fri, Mar 4, 2011, subscribers received the following message from Gizmo5, indicating that the service would be terminated.
There was no indication made if the service would be revived in another form, or if there would be similar functionality added to any of Google's current telephony offerings. On the morning of April 4, service was finally cut.[ vague ]
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE).
Telephony is the field of technology involving the development, application, and deployment of telecommunication services for the purpose of electronic transmission of voice, fax, or data, between distant parties. The history of telephony is intimately linked to the invention and development of the telephone.
Speex is an audio compression codec specifically tuned for the reproduction of human speech and also a free software speech codec that may be used on voice over IP applications and podcasts. It is based on the code excited linear prediction speech coding algorithm. Its creators claim Speex to be free of any patent restrictions and it is licensed under the revised (3-clause) BSD license. It may be used with the Ogg container format or directly transmitted over UDP/RTP. It may also be used with the FLV container format.
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for voice calls for the delivery of voice communication sessions over Internet Protocol (IP) networks, such as the Internet.
Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints such as customary telephone sets, destinations on the public switched telephone network (PSTN) and devices or services on voice over Internet Protocol (VoIP) networks. Its name comes from the asterisk (*) symbol for a signal used in dual-tone multi-frequency (DTMF) dialing.
Google Talk was an instant messaging service that provided both text and voice communication. The instant messaging service was variously referred to colloquially as Gchat, Gtalk, or Gmessage among its users.
Skype for Business Server is real-time communications server software that provides the infrastructure for enterprise instant messaging, presence, VoIP, ad hoc and structured conferences and PSTN connectivity through a third-party gateway or SIP trunk. These features are available within an organization, between organizations and with external users on the public internet or standard phones.
A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet. This is in contrast to a standard phone which uses the traditional public switched telephone network (PSTN).
SipXecs is a free software enterprise communications system. It was initially developed by Pingtel Corporation in 2003 as a voice over IP telephony server located in Boston, MA. The server was later extended with additional collaboration capabilities as part of the SIPfoundry project. Since its extension, sipXecs now acts as a software implementation of the Session Initiation Protocol (SIP), making it a full IP-based communications system.
QuteCom was a free-software SIP-compliant VoIP client developed by the QuteCom community under the GPL-2.0-or-later license. It allows users to speak to other users of SIP-compliant VoIP software at no cost. It also allows users to call landlines and cell phones, send SMS and make video calls. None of these functions are tied to a particular provider, allowing users to choose among any SIP provider.
This is a comparison of voice over IP (VoIP) software used to conduct telephone-like voice conversations across Internet Protocol (IP) based networks. For residential markets, voice over IP phone service is often cheaper than traditional public switched telephone network (PSTN) service and can remove geographic restrictions to telephone numbers, e.g., have a PSTN phone number in a New York area code ring in Tokyo.
Jitsi is a collection of free and open-source multiplatform voice (VoIP), video conferencing and instant messaging applications for the Web platform, Windows, Linux, macOS, iOS and Android. The Jitsi project began with the Jitsi Desktop. With the growth of WebRTC, the project team focus shifted to the Jitsi Videobridge for allowing web-based multi-party video calling. Later the team added Jitsi Meet, a full video conferencing application that includes web, Android, and iOS clients. Jitsi also operates meet.jit.si, a version of Jitsi Meet hosted by Jitsi for free community use. Other projects include: Jigasi, lib-jitsi-meet, Jidesha, and Jitsi.
Google Voice is a telephone service that provides a U.S. phone number to Google Account customers in the U.S. and Google Workspace customers in Canada, Denmark, France, the Netherlands, Portugal, Spain, Sweden, Switzerland, the United Kingdom and the contiguous United States. It is used for call forwarding and voicemail services, voice and text messaging, as well as U.S. and international calls. Calls are forwarded to the phone number that each user must configure in the account web portal. Users can answer and receive calls on any of the phones configured to ring in the web portal. While answering a call, the user can switch between the configured phones. Subscribers in the United States can make outgoing calls to domestic and international destinations. The service is configured and maintained by users in a web-based application, similar in style to Google's email service Gmail, or Android and iOS applications on smartphones or tablets.
Empathy was an instant messaging (IM) and voice over IP (VoIP) client which supported text, voice, video, file transfers, and inter-application communication over various IM communication protocols.
Wideband audio, also known as wideband voice or HD voice, is high definition voice quality for telephony audio, contrasted with standard digital telephony "toll quality". It extends the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech. The range of the human voice extends from 100 Hz to 17 kHz but traditional, voiceband or narrowband telephone calls limit audio frequencies to the range of 300 Hz to 3.4 kHz. Wideband audio relaxes the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz. In addition, some wideband codecs may use a higher audio bit depth of 16 bits to encode samples, also resulting in much better voice quality.
A softphone is a software program for making telephone calls over the Internet using a general purpose computer rather than dedicated hardware. The softphone can be installed on a piece of equipment such as a desktop, mobile device, or other computer and allows the user to place and receive calls without requiring an actual telephone set. Often, a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a handset, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC or with a USB phone.
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Acrobits is a privately owned software development company creating VoIP Clients for mobile platforms, based in Prague, Czech Republic.